Move VoiceEngineObserver into AudioSendStream so that we detect typing noises and return properly i…

webrtc/call/call_unittest.cc

Index: webrtc/call/call_unittest.cc
diff --git a/webrtc/call/call_unittest.cc b/webrtc/call/call_unittest.cc
index 9819b538f85c51e66edef33b7b9059a6febf502d..672195311d3c853f0029974267a8889d61b9b7fd 100644
--- a/webrtc/call/call_unittest.cc
+++ b/webrtc/call/call_unittest.cc
@@ -12,6 +12,7 @@
 
 #include "testing/gtest/include/gtest/gtest.h"
 
+#include "webrtc/audio_state.h"
 #include "webrtc/call.h"
 #include "webrtc/test/fake_voice_engine.h"
 
@@ -19,8 +20,11 @@ namespace {
 
 struct CallHelper {
   CallHelper() : voice_engine_(new webrtc::test::FakeVoiceEngine()) {
+    webrtc::AudioState::Config audio_state_config;
+    audio_state_config.voice_engine = voice_engine_.get();
+    audio_state_.reset(webrtc::AudioState::Create(audio_state_config));
     webrtc::Call::Config config;
-    config.voice_engine = voice_engine_.get();
+    config.audio_state = audio_state_.get();
     call_.reset(webrtc::Call::Create(config));
   }
 
@@ -28,6 +32,7 @@ struct CallHelper {
 
  private:
   rtc::scoped_ptr<webrtc::test::FakeVoiceEngine> voice_engine_;
+  rtc::scoped_ptr<webrtc::AudioState> audio_state_;
   rtc::scoped_ptr<webrtc::Call> call_;
 };
 }  // namespace